[rhythmbox] xfade: use appsrc to produce silence rather than audiotestsrc

[Jonathan Matthew <jmatthew src gnome org>]

commit 039da368a66544ca578a0dd121a2687f63ea6125
Author: Jonathan Matthew <jonathan d14n org>
Date:   Wed Jan 6 22:59:53 2016 +1000

    xfade: use appsrc to produce silence rather than audiotestsrc
    
    This works around what appears to be a bug in pulsesink, where it
    doesn't send any data to pulseaudio when it receives a GAP buffer,
    which means the stream timestamps get out of sync.
    
    https://bugzilla.gnome.org/show_bug.cgi?id=757225

 backends/gstreamer/rb-player-gst-xfade.c |   72 +++++++++++++++++++++++++++---
 1 files changed, 65 insertions(+), 7 deletions(-)
---
diff --git a/backends/gstreamer/rb-player-gst-xfade.c b/backends/gstreamer/rb-player-gst-xfade.c
index 739e4c8..01bca3b 100644
--- a/backends/gstreamer/rb-player-gst-xfade.c
+++ b/backends/gstreamer/rb-player-gst-xfade.c
@@ -190,6 +190,8 @@ static gboolean start_sink (RBPlayerGstXFade *player, GError **error);
 static gboolean stop_sink (RBPlayerGstXFade *player);
 static void maybe_stop_sink (RBPlayerGstXFade *player);
 
+static gboolean silencesrc_push (RBPlayerGstXFade *player);
+
 GType rb_xfade_stream_get_type (void);
 GType rb_xfade_stream_bin_get_type (void);
 
@@ -243,6 +245,7 @@ struct _RBPlayerGstXFadePrivate
        /* probably don't need to store pointers to these either */
        GstElement *pipeline;
        GstElement *outputbin;
+       GstElement *silencesrc;
        GstElement *silencebin;
        GstElement *adder;
        GstElement *capsfilter;
@@ -279,6 +282,9 @@ struct _RBPlayerGstXFadePrivate
 
        guint bus_idle_id;
        GList *idle_messages;
+
+       char silence_buffer[1024];
+       guint silence_idle_id;
 };
 
 
@@ -2859,7 +2865,7 @@ stream_volume_changed (GObject *element, GParamSpec *pspec, RBPlayerGstXFade *pl
  *
  * outputcaps = audio/x-raw,channels=2,rate=44100,format=S16LE
  * outputbin = outputcaps ! volume ! filterbin ! audioconvert ! audioresample ! tee ! queue ! audiosink
- * silencebin = audiotestsrc wave=silence ! outputcaps
+ * silencebin = appsrc ! outputcaps
  *
  * pipeline = silencebin ! adder ! outputbin
  *
@@ -2923,6 +2929,9 @@ start_sink_locked (RBPlayerGstXFade *player, GList **messages, GError **error)
                return FALSE;
        }
 
+       /* give the silence bin some data so it can preroll */
+       silencesrc_push (player);
+
        /* now wait for everything to finish */
        waiting = TRUE;
        bus = gst_element_get_bus (GST_ELEMENT (player->priv->pipeline));
@@ -3185,12 +3194,50 @@ stop_sink (RBPlayerGstXFade *player)
        return TRUE;
 }
 
+static void
+silencesrc_free_buffer(gpointer d)
+{
+}
+
+static gboolean
+silencesrc_push (RBPlayerGstXFade *player)
+{
+       GstBuffer *buffer;
+       GstFlowReturn ret;
+
+       buffer = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
+                                             player->priv->silence_buffer,
+                                             sizeof(player->priv->silence_buffer),
+                                             0,
+                                             sizeof(player->priv->silence_buffer),
+                                             silencesrc_free_buffer,
+                                             NULL);
+       g_signal_emit_by_name (player->priv->silencesrc, "push-buffer", buffer, &ret);
+       gst_buffer_unref (buffer);
+
+       return (ret == GST_FLOW_OK);
+}
+
+static void
+silencesrc_need_data_cb (GstElement *appsrc, guint size, RBPlayerGstXFade *player)
+{
+       if (player->priv->silence_idle_id == 0)
+               player->priv->silence_idle_id = g_idle_add ((GSourceFunc) silencesrc_push, player);
+}
+
+static void
+silencesrc_enough_data_cb (GstElement *appsrc, RBPlayerGstXFade *player)
+{
+       if (player->priv->silence_idle_id != 0) {
+               g_source_remove (player->priv->silence_idle_id);
+               player->priv->silence_idle_id = 0;
+       }
+}
 
 static gboolean
 create_sink (RBPlayerGstXFade *player, GError **error)
 {
        const char *try_sinks[] = { "gsettingsaudiosink", "gconfaudiosink", "autoaudiosink" };
-       GstElement *audiotestsrc;
        GstElement *audioconvert;
        GstElement *audioresample;
        GstElement *capsfilter;
@@ -3312,8 +3359,19 @@ create_sink (RBPlayerGstXFade *player, GError **error)
 
        /* create silence bin */
        player->priv->silencebin = gst_bin_new ("silencebin");
-       audiotestsrc = gst_element_factory_make ("audiotestsrc", "silence");
-       g_object_set (audiotestsrc, "wave", 4, NULL);
+
+       /*
+        * audiotestsrc is the sensible thing to use here, except that with the
+        * silent waveform it produces buffers with the GAP flag set, which currently
+        * cause pulsesink to screw up.
+        *
+        * to get around this, for now we produce silence using an appsrc instead.
+        */
+       player->priv->silencesrc = gst_element_factory_make ("appsrc", "silencesrc");
+       g_object_set (player->priv->silencesrc, "caps", caps, "format", GST_FORMAT_TIME, NULL);
+
+       g_signal_connect (player->priv->silencesrc, "need-data", G_CALLBACK (silencesrc_need_data_cb), 
player);
+       g_signal_connect (player->priv->silencesrc, "enough-data", G_CALLBACK (silencesrc_enough_data_cb), 
player);
 
        audioconvert = gst_element_factory_make ("audioconvert", "silenceconvert");
 
@@ -3321,7 +3379,7 @@ create_sink (RBPlayerGstXFade *player, GError **error)
        g_object_set (capsfilter, "caps", caps, NULL);
        gst_caps_unref (caps);
 
-       if (audiotestsrc == NULL ||
+       if (player->priv->silencesrc == NULL ||
            audioconvert == NULL ||
            capsfilter == NULL) {
                g_set_error (error,
@@ -3332,11 +3390,11 @@ create_sink (RBPlayerGstXFade *player, GError **error)
        }
 
        gst_bin_add_many (GST_BIN (player->priv->silencebin),
-                         audiotestsrc,
+                         player->priv->silencesrc,
                          audioconvert,
                          capsfilter,
                          NULL);
-       if (gst_element_link_many (audiotestsrc,
+       if (gst_element_link_many (player->priv->silencesrc,
                                   audioconvert,
                                   capsfilter,
                                   NULL) == FALSE) {